Speech transmission by means of an IP protocol in local networks, for example in a VoIP system as a supplement to or substitute for an existing communication system (PBX) or as Internet telephony in the long-distance zone is currently still not very widespread. However, it represents a very promising communication method for the future. In combination with other IP-based services, it permits novel, interesting forms of communication.
However, a range of organizational as well as technical problems remain to be solved in the interests of a wide penetration of VoIP.
Particularly in networks without QoS (Quality of Service) mechanisms, the complex interplay of network, communication protocol, operating system and hardware means that speech transmission over IP networks does not deliver constant speech quality. A method is therefore required for assessing the speech quality in order to log the latter (if appropriate, even to signal it) and/or to use it in a subsequent process as control information for adaptive/self-optimizing systems (bandwidth control, and endpoints, alternative route selection, for example at gateways, etc.).
The method should meet the following requirements:                no additional loading of the network by test signals,        assessment of the quality of real speech connections, not the quality of test signals,        automatic determination of the quality,        real time capability of the method (determination of the quality during operation),        simplicity of the method (no complicated calculations; low processor loading),        continuous applicability (running generation of measured data),        full-area applicability (each end point should be taken into account)        transparency for the user of a Voice over IP system (no impairment of the function).        
Previous solutions for determining the speech quality can be classified approximately as follows:                a) assessment of network parameters (round-trip delay, etc.) without concrete reference to the particular features of the real time communication; therefore unsuitable for providing adequate information.        b) application layer measurements (PSQM, etc.) in which a special signal is fed in at the transmitter for transmitting via the channel to be examined, and is compared with the signal received at the receiver with the assistance of complex mathematical models, inter alia in order to take account of the physiological properties of the human ear. On the one hand, an additional network load is generated by test samples, while on the other hand the calculation algorithms consume a not inconsiderable portion of the processor power. Appropriate test equipment is indeed available, but can only be used for individual measurements (not least because of the high outlay on apparatus and funds).        c) test series with the participation of testing staff in order to determine the speech quality (MOS). This method is unsuitable from the very start for the continuous application in real installations and very expensive in terms of staff. It can therefore be stated that although methods are available for determining speech quality, none of them meets the above-named requirements.        